Introduction

This release note applies to all 8.5 releases of Genesys WebRTC Gateway.

Links in the Contents section enable you to access information
regarding a specific release. Releases are listed by version
number rather than in date order. For this reason, a recent
release may be listed after earlier releases, if the version
number is lower. Except when otherwise noted in the information
for a specific release, each release includes all of the features
and corrections that were introduced for the applicable operating
system at earlier dates, regardless of the version numbering.

The Known Issues and Recommendations section is a cumulative
list. It includes information on when individual items were
found and, if applicable, corrected.

The Corrections and Modifications
section for each release may list additional issues that were
corrected without first being documented as Known Issues.

The Discontinued Support section is cumulative. It lists
functionality that is no longer supported and the release
number of the version in which support was discontinued.

Use of Third-Party Software

Genesys follows applicable third-party redistribution policies to the extent
that Genesys solutions utilize third-party functionality. For specific
information on any third-party software used in this product, see the Legal Notices for WebRTC. Please contact your Genesys Customer Care representative if you have any
questions.

Release Number 8.5.210.75 [08/03/2018] – Hot Fix

Supported Operating Systems

The operating systems supported by this release are listed in the Contents,
above.

New in This Release

There are no restrictions for this release. This release contains no new features
or functionality.

Corrections and Modifications

This release also includes the following correction and modification:

WebRTC Gateway now correctly processes the media streams by performing SRTP renegotiation when retrieving a video call from hold. Previously, retrieving a video call from hold might fail because WebRTC Gateway did not perform SRTP renegotiation when the remote key was changed and SSRC was unchanged. (MWA-687)

Release Number 8.5.208.71 [05/12/2017] – General

Supported Operating Systems

The operating systems supported by this release are listed in the Contents,
above.

New in This Release

There are no restrictions for this release. This release contains no new features or functionality.

Corrections and Modifications

This release also includes the following corrections and modifications:

Session re-negotiations now work correctly in Firefox 55 Nightly, with a=setup:actpass set in offer SDPs by the gateway. Previously, the setup attribute was not set during re-negotiations, resulting in failure. (MWA-590)

Media stream is now established correctly when making a call using Firefox version 51 or 52. Previously, this did not work due to incompatible ciphers, and is now fixed by updating the version of the OpenSSL library. (MWA-584)

Release Number 8.5.208.16 [01/31/2017] – Hot Fix

Supported Operating Systems

The operating systems supported by this release are listed in the Contents,
above.

New in This Release

This is a hot fix for this product. This section describes new features that
were introduced in this release of Genesys WebRTC Gateway.

A boolean option, [rsmp] web-disable-sdes, has been added with a default value of true, which disables SDES-SRTP in initial SDP offers to the web client. With a value of true, the RTP profile in an initial offer SDP from the gateway to a web client will use "UDP/TLS/RTP/SAVPF" instead of "RTP/SAVPF", adhering to the WebRTC standard. Even with a value of true, incoming offers with SDES-SRTP will still be accepted, though the use of SDES-SRTP is obsolete and discouraged. (MWA-575)

The Genesys WebRTC Gateway now has G.722 audio codec enabled by default. Note that G.722 codec is given higher priority than G.711 codecs by the browsers as well as the WebRTC gateway, therefore G.722 may be chosen over G.711 during call establishment. (MWA-98)

Corrections and Modifications

This release also includes the following corrections and modifications:

An ICE connection is now established between Mozilla Firefox and WebRTC Gateway, when the Gateway is in a private network and is using a STUN Server to find its server reflexive address. (MWA-536)

The G.722 audio codec is now enabled on the WebRTC Gateway by default. (MWA-98)

An audio-only call can now be resumed after video-on-hold is played to the browser client, during a hold by the peer. Previously in Mozilla Firefox, resuming the call did not work. (MWA-576)

Release Number 8.5.206.25 [04/29/2016] – Hot Fix

Supported Operating Systems

The operating systems supported by this release are listed in the Contents,
above.

New in This Release

This is a hot fix for this product. This section describes new features that
were introduced in this release of Genesys WebRTC Gateway.

The following configuration option has been added, which helps
work around some renegotiation issues with Firefox. When an older version
of Firefox without bundle support is used (version 37 and lower), set this
option to false. (MWA-547)

Corrections and Modifications

This release also includes the following corrections and modifications:

In WebRTC Gateway, the client message queue to the browser is now cleared on
a new call if a problem is detected. Previously, the queue might have become
full due to some client issue, which caused any new message going to the client
to be dropped. (MWA-528)

When upgrading a WebRTC call from audio only to include video in the Firefox
browser, media bundle will now be used, if it is enabled on the platform. Previously,
media bundle could not be used in this scenario. (MWA-518)

Release Number 8.5.204.88 [10/23/2015] – Hot Fix

Supported Operating Systems

The operating systems supported by this release are listed in the Contents,
above.

New in This Release

This is a hot fix for this product. This release contains no new features
or functionality.

Corrections and Modifications

This release also includes the following corrections and modifications:

A new SIP INVITE that is immediately preceded by an SDP negotiation
is now handled correctly by appropriate queuing. Previously in this scenario,
the SIP INVITE was rejected with the error 486 Busy Here.
(MWA-499)

WebRTC Gateway now checks configured media codecs’ dynamic payload numbers
and, if a number is out of range, it ignores the configured value and selects
a valid value. Previously, the Gateway terminated when a dynamic payload number
that was configured in the codecs configuration parameter was out of range.
(MWA-500)

The size of the message queue to the web client has been increased in order
to avoid related failures. (MWA-505)

The media-level tags that are used for media bundle support in an SDP Answer
now match the tags in the SDP Offer. Previously, these tags were
hardcoded to the tags used by Chrome. This hardcoding caused issues with Firefox,
which recently added media BUNDLE support. (MWA-506)

The Datagram Transport Layer Security (DTLS) establishment between WebRTC Gateway
and Firefox version 38 and higher now works correctly when Firefox is in an
active role, that is, when Firefox is the callee. (MWA-507)

You can now use the rsmp.sip-proxy configuration parameter to
specify two SIP Server addresses, separated by a comma, which will be used for
High Availability. Only one of the SIP Servers will be active at any one time
and, when a request to it times-out, the Gateway will start using the other
SIP Server. Also, you can use the configuration parameter rsmp.sip-proxy-srv
to configure the SRV address of the SIP Server. When a SIP request arrives from
this SRV address, the response is sent to the currently active SIP Server address.
(MWA-508)

Release Number 8.5.202.44 [12/12/2014] – Hot Fix

Supported Operating Systems

The operating systems supported by this release are listed in the Contents,
above.

New in This Release

This is a hot fix for this product. This release contains no new features
or functionality.

Corrections and Modifications

This release also includes the following corrections and modifications:

A conferencing issue no longer occurs when using WebRTC Gateway with Chrome
version 36 or later. Previously, the following issue occurred:

Inbound call was answered by Agent1.

Agent1 consulted with Agent2.

Agent1 completed conference.

Agent1 could not hear Caller and Agent2.

(MWA-456)

Possible issues in Chrome with SRTP are now avoided during hold and resume
operations when music on hold is enabled. Previously, this may have caused issues
when the media source changed on the SIP side, as delayed packets from the old
media source may have been forwarded to the web client. (MWA-454)

DTLS establishment in the WebRTC Gateway now happens quickly and consistently.
Previously, it could be arbitrarily delayed when the Gateway was running on
a Linux system. (MWA-464)

Release Number 8.5.201.95 [10/06/2014] – General

Supported Operating Systems

The operating systems supported by this release are listed
in the Contents, above.

New in This Release

There are no restrictions for this release. This section describes new features
that were introduced in this release of Genesys WebRTC Gateway.

The Genesys WebRTC Gateway now supports adding video to an audio-only call.

The Genesys WebRTC Gateway now supports remote CTI control by providing
the SIP extensions event package known as the BroadSoft SIP extensions.

Corrections and Modifications

This release also includes the following corrections and modifications:

When video is disabled and then re-enabled from the SIP-side during a call
using DTLS and media-bundle, the video now plays correctly on the web-side.
Previously, the video would not play correctly on the web-side under these circumstances.
Also, if you carry out hold and resume operations with music-on-hold while using
DTLS, these operations now work correctly. (MWA-437)

You can now use the web-ice-addresses configuration parameter
to specify multiple local IP addresses for use by the ICE library. The WebRTC
Gateway also supports HTTP OPTIONS. (MWA-438)

When using the Chrome web browser, if you re-enable video after disabling it,
Chrome receives video from the peer, as expected. However, it will not send
video to the peer, due to
WebRTC issue
2136. (MWA-442)

Release Number 8.5.201.30 [07/18/2014] – General

Supported Operating Systems

The operating systems supported by this release are listed
in the Contents, above.

New in This Release

There are no restrictions for this release. This section describes new features
that were introduced in this release of Genesys WebRTC Gateway.

Support for third-party call control (3pcc) basic functions and two-step procedures

Corrections and Modifications

This release also includes the following corrections and modifications:

Video media are now correctly played when a callee sends a new mid-call
offer. Previously, video media were not always played when a new offer was
sent. ICE checks are also now only carried out once during session
renegotiation, unless more than one ICE check is required. (MWA-390)

New media streams (for example, video) can now be added to a call
when using DTLS-SRTP. Support for a new PeerConnection by
re-establishing DTLS for all media streams has also been added. This is needed
with browsers that do not support call renegotiation, such as Firefox (as
explained in Firefox
Bug
857115). (MWA-392)

Corrections and Modifications

Known Issues and Recommendations

This section provides the latest information on known issues and
recommendations associated with this product.

Upgrading a video-only call to an audio plus video call currently does not work. (MWA-594).

Found In: 8.5.208.71

Fixed In:

Firefox has now added support for session renegotiation. However, the default
behavior in the gateway and the Genesys WebRTC JavaScript API is to create a
new PeerConnection for each renegotiation. You can take advantage
of the Firefox renegotiation support without having to create new PeerConnections
by doing the following:

Set the configuration option rsmp.new-pc-support to 0
(the default value is 1). Note that you will need to add this
option in Configuration Manager.

Set a corresponding option in the Genesys WebRTC JavaScript API by calling
the Grtc.Client method setRenewSessionOnNeed(false).

Be aware, however, that the Firefox renegotiation support has not been well
tested, and is disabled in the platform by default.

The Chrome browser, since version 47, supports user media operations only from
secure origin. If you are using Genesys WebRTC service with Chrome browser version
47 or higher, you must use HTTPS for the application server, as well as the
WebRTC Gateway. To do this, configure the WebRTC Gateway and set up the application
server with the appropriate security certificates as described in the Genesys
WebRTC Service Deployment Guide.

The following conferencing issue occurs when using WebRTC Gateway with Chrome
version 36 or 37 (latest stable version):

Inbound call is answered by Agent1.

Agent1 consults with Agent2.

Agent1 completes conference.

Agent1 cannot hear Caller and Agent2.

Agent2 and Caller can hear each other, and they can also hear Agent1.

To work around this issue, Agent1 can hold and then resume the call, and all
three parties will be able to hear each other.

(MWA-456)

Found In: 8.5.201.95

Fixed In: 8.5.202.44

Google Chrome has an outstanding issue that affects the WebRTC Gateway:

When you use the Genesys WebRTC Service and the Genesys Voice Platform Media
Control Platform (MCP) on Red Hat Enterprise Linux (RHEL), Genesys recommends
that MCP be run on a server that is separate from the server running the
WebRTC Service, for two reasons:

MCP performance on RHEL 5.x is considerably better than on RHEL 6.x. However,
the WebRTC Gateway only works on RHEL 6.x.

MCP consumes the majority of the resources of the server it is hosted on,
even under RHEL 5.x. Because of this, Genesys recommends—regardless of
whether WebRTC is running in your environment—that MCP be placed on its
own, separate machine.

The WebRTC Gateway displays a memory leak while running under Windows, due to an
issue with the LibNice/GLib third-party library that is used by the Gateway.
The amount of the leak varies with the call scenario being used. For example,
for browser-to-browser call scenarios the observed leak is approximately 70
bytes per audio-only call and approximately 800 bytes per audio-video call.

If you are deploying the WebRTC Gateway in a Windows environment, Genesys
recommends that you monitor memory and be prepared to restart the Gateway
in cases where it becomes necessary to do so. (MWA-321)

Note that the WebRTC Gateway normally supports only the two most recent versions
of supported browsers.

Discontinued Support

This section documents features that are no longer supported in this software. This cumulative list is in release-number order with the most recently discontinued features at the top of the list. For more information on discontinued support for operating environments and databases, see Discontinued Support in the Genesys Supported Operating Environment Reference Guide.

Internationalization

Additional Information

Additional information on Genesys Telecommunications Laboratories, Inc. is
available on our Customer Care website.
The following documentation also contains information about this software. Please
consult the Deployment Guide first.

The
Genesys
WebRTC Service Deployment Guide gives you the information that you need to
get started with Genesys WebRTC Service. This document contains product overview
information, as well as deployment details.

Note: For the DVD, the New Documents on this DVD page indicates
the production date for that disc. Due to disc production schedules, documentation
on the Genesys Documentation website may be more up-to-date than what is available
on disc immediately after a product is released or updated. To determine the
version of a document, check the version number that is located on the second
page in PDFs or on the About This File topic in Help files.